ABSTRACT

This thesis provides a description of transmission and reception of a narrow-

band signal using a soundcard for pilot tone detection in spectrum sensing. The

energy of pilot tone and the signal around bandwidth of 25kHz are calculated and

the energy of the spectrum is compared with test statistics. The method is based

on noise and signal plus noise statistics. The energy detection of a random signal

with white Gaussian noise is reduced to the sum of squares of statistically inde-

pendent Gaussian random samples. When the signal is absent, the statistics of the

energy is chi-square distributed. If the signal is present, statistics has non-central

chi-square distribution with a non-centrality parameter which is a function of the

signal strength. The probability density function (pdf) of energy of noise and signal

plus noise is analytically calculated. A phase locked loop(PLL) is used to synchronize

the local oscillator with the carrier of the received signal. Designed and implemented

highpass and lowpass filter using bilinear transform as a component to use in the

PLL. Probability of error and probability of false alarm for the signal plus noise are

calculated. Obtained the optimum value of threshold the decreasing amplitude for

different observation time.

 TABLE OF CONTENTS

CHAPTER

LIST OF NOTATIONS . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1

LIST OF UNITS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1

1 INTRODUCTION 1

1.1 ATSC Standard . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3

1.2 Background . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4

1.3 Problem Statement . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9

2 BASEBAND SIGNAL PROCESSING AND ENERGY DETECTION 11

2.1 Statistics of Low-pass Filtered White Noise . . . . . . . . . . . . . . . 11

2.2 Statistics of Low-pass Filtered Signal + Noise . . . . . . . . . . . . . 15

2.3 Energy Based Signal Detection . . . . . . . . . . . . . . . . . . . . . 20

3 DETECTION OF BASEBAND SIGNAL 26

3.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26

3.2 Problem Statement . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27

3.3 Signal Processing of Transmitted Signal . . . . . . . . . . . . . . . . . 28

3.4 Signal Processing of Received Signal . . . . . . . . . . . . . . . . . . 34

v
 3.5 Energy Statistics of Narrowband Noise . . . . . . . . . . . . . . . . . 41

3.6 Energy Statistics of Narrowband Signal + Noise . . . . . . . . . . . . 43

3.7 Detection of a Narrow-band Signal in Noise . . . . . . . . . . . . . . 45

4 CONCLUSION 49

BIBLIOGRAPHY 51

vi
 LIST OF TABLES

Table

2.1 The probability of detection ( Pdet ) and the probability of false alarm

( Pf a ) for SN R = 4 . . . . . . . . . . . . . . . . . . . . . . . . . . . 24

2.2 The probability of detection ( Pdet ) and the probability of false alarm

( Pf a ) for SN R = 1 . . . . . . . . . . . . . . . . . . . . . . . . . . . 25
 LIST OF FIGURES

Figure

2.1 Interpolation of discrete white noise using low-pass filter . . . . . . . 12

2.2 Probability density function of noise energy: (A) 2M = 4, (B) 2M =

6, (C) 2M = 8, (D) 2M = 10 . . . . . . . . . . . . . . . . . . . . 15

2.3 Probability density function of signal + noise . . . . . . . . . . . . . 17

2.4 Pdf of signal plus noise energy: (A) 2M = 2, (B) 2M = 4, (C)

2M = 8. Continuous line is FFT based estimation of fQ (q). Symbols

are simulated results. . . . . . . . . . . . . . . . . . . . . . . . . . . 19

2.5 Optimum threshold for 50% duty cycle for M = 4 , A2 = 3 and σ 2 = 1 22

2.6 The error probability and the probability of false alarm for SN R = 4 23

2.7 The error probability and the false alarm probability of signal plus

noise for SN R = 1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24

3.1 Spectrum of VSB (http://www.8vsb.com) . . . . . . . . . . . . . . . 27

3.2 Passband spectrum of the information signal . . . . . . . . . . . . . . 30

3.3 Modulated suppressed carrier passband signal . . . . . . . . . . . . . 30

3.4 Modulated signal when modulation index, m ≤ 1 . . . . . . . . . . . 31

3.5 Modulated signal when modulation index, m ≥ 1 . . . . . . . . . . . 31

3.6 Baseband signal . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33

3.7 Transmitted signal . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33

3.8 Received signal after demodulation . . . . . . . . . . . . . . . . . . . 34

3.9 Block diagram of phase locked loop . . . . . . . . . . . . . . . . . . . 36

3.10 Theta vs N . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37

3.11 Down-converted signal . . . . . . . . . . . . . . . . . . . . . . . . . . 41

3.12 Transfer function of lowpass filter . . . . . . . . . . . . . . . . . . . . 42

3.13 Pdf of noise energy in the intermediate frequency: (A) 2M = 2,

(B) 2M = 4, (C) 2M = 8, (D) 2M = 10 . . . . . . . . . . . . . . . 43

3.14 Pdf of the signal plus noise(Analytical and simulation Method): (A) 2M =

2, (B) 2M = 4, (C) 2M = 6, (D) 2M = 8 . . . . . . . . . . . . . . 45

3.15 Pe versus Threshold Amplitude for fixed window size: (A) 2M = 2,

(B) 2M = 4 , (C) 2M = 6, (D) 2M = 8, (E) 2M = 10 . . . . . . . 47

3.16 Pe versus SN R for fixed window size 2M = 2 . . . . . . . . . . . . . 47

ix
 CHAPTER 1

INTRODUCTION

The increasing demand for wireless network access and crowding of the un-

licensed spectral bands has made the authorities to look for new ways to allocate

spectral resources. Currently service providers are faced with a situation where they

require a larger fraction of the available bandwidth to improve the quality of ser-

vice. Under the current system of spectrum allocation, rigid partitioning has created

some vastly underutilized spectral bands. A proposed way to reuse this underutilized

spectrum is through dynamically accessing the unused spectrum in an opportunistic

manner. But there are fundamental tradeoffs to primary users who may have al-

ready licensed the spectrum. The secondary users who opportunistically access the

spectrum should be able to detect the presence of primary users through continuous

spectrum sensing. Spectrum sensing is a challenging part of the dynamical chan-

nel access problem. The question being, how effectively can secondary users sense

and adapt to the presence of a signal in a spectral channel. Lack of robustness in

an uncertain noise environment is a typical bottleneck of current spectrum sensing

strategies for detecting the primary signal. The objective is to make robust and

reliable spectrum sensing possible in a weak signal environment.

 2

Cognitive radio systems should have the ability to sense vacant spectrum and

use it opportunistically while preserving priority for primary users. However, real

world uncertainties such as channel fading and shadowing makes the spectrum sens-

ing with high sensitivity and robustness difficult. Detecting the presence or absence

of weak signals is quite challenging. Fading is the change in attenuation of a carrier

modulated telecommunication signal in certain propagation media. The fading can

vary with time, geographical position and frequency of the signal. Most of the time,

fading is caused by multipath effects. In that the transmitted signal can travel by

multiple paths to the receiver. Multi-path affects robust signal detection and the

received signal to noise ratio (SNR) level. Introduction of a deterministic pilot tone

signal can aid in ones ability to detect the presence of a signal in noise. However

phase synchronization is challenging problem when one uses pilot tone detection

[1][2].

Different regions of the spectrum are used for different radio transmission tech-

nologies and applications. The Federal Communication Commission(FCC) ruled that

unlicensed devices can use white space in the frequency spectrum if they can guaran-

teed that they will not interfere with assigned broadcasts. These requirements render

significant challenges to the unlicensed spectrum access. Spectrum sensors makes a

series of measurements and then utilize the measurements for making a decision on

the channel being occupied or not.

 3

1.1 ATSC Standard

The Advanced Television Systems Committee (ATSC) developed standards

for digital television transmission over terrestrial, cable, and satellite networks. The

Advanced Television Systems Committee Inc., is an international, non profit orga-

nization developing voluntary standards for digital television. ATSC is working to

coordinate television standards among different communications media focusing on

digital television, interactive systems and broadband multimedia communications.

ATSC is also developing digital television implementation strategies and presenting

educational seminars on the ATSC standards. ATSC digital TV standards include

digital high definition television (HDTV), standard definition television (SDTV),

data broadcasting, multichannel surround-sound audio, and satellite direct-to-home

broadcasting. ATSC standards were adopted by the FCC in the United States in

1996. It was revised in 2009. ATSC replaced much of the analog NTSC television

system in the United States on June 12, 2009. ATSC signals are designed to use the

same 6 MHz bandwidth as the analog NTSC signal it replaced. Digital video and au-

dio signals are compressed and multiplexed, and the transport stream is modulated

in different ways depending on the method of transmission. Terrestrial broadcasters

use 8-VSB modulation that can transfer at a maximum rate of 19.39Mbits/s. This

rate is sufficient to carry several video and audio programs.

 4

1.2 Background

In the presence of noise, signal detection is not possible when its amplitude

falls below a threshold value. The problem of detecting whether a frequency band

is being used by a primary user in a weak signal environment is the bottleneck in

providing dynamic spectrum access. In such a case the signal to noise power ratio is

small. Adding a pilot tone to the transmitting signal enhances one’s ability to detect

the presence of a signal. Pilot tones are single frequency signals transmitted over

a communications channel for supervising and reference purposes. But this benefit

comes at the cost of perfect synchronization required to demodulate the signal.

The opening of the television spectrum to opportunistic access has generated

interest in the development of effective sensing mechanisms to detect the presence of

television signals. Successful approaches will allow one to access those bands when

they are not used in TV transmission. There are several classifications of sensing

techniques for a weak TV signal. First, a sensing technique can be classified as either

signal specific or as a blind sensing technique. A blind sensing technique does not

rely on features of a specific signal type. But a signal specific sensing technique is

based on features of a specific signal type. Another classification is the coarse sensing

technique and fine sensing technique. If a signal is detected it implies that channel

is occupied. If the channel is not occupied then fine sensing can be used to sense for

weaker signals. There are three blind sensing techniques. They are energy detection,

eigenvalue based detection and the multi-resolution technique.

Energy detection is a non-coherent detection method. As such the receiver does

 5

not require knowledge of the features of the primary user signal. In this method an

energy detector is used to measure the energy of the signal over a specified time

interval. The measured energy is compared to a prescribed threshold energy ampli-

tude. This sensing technique relies on apriori knowledge of the noise power. If the

estimated energy of the received signal is greater than the threshold amplitude, one

concludes that a signal generated by a primary user is present. Low computational

complexity makes the method popular. Since it does not depend on the temporal

characteristics of the signal, the method can be applied to the detection of a variety

of transmitted signals [3][4][5]. However the method is not robust at low SNR levels.

The method of eigenvalue based sensing is explained in [3]. The sensor cal-

culates estimates of the autocorrelation function of the received signal. Statistics

of an RF signal is likely to be different than that of noise. That difference will be

characterized by the eigenvalue distributions of the sample covariance matrix. The

correlation matrix is then transformed by a whitening matrix Q which corrects for

the frequency response of the receiver. Sensing algorithms are based on either the

maximum to minimum eigenvalue ratio or the average to minimum eigenvalue ratio

[6] to establish the decision amplitude threshold.

The total sensing time can be reduced using multi-resolution spectrum sensing.

First bandwidth is sensed using a course resolution. Fine resolution sensing is then

performed over a small range of frequencies. In order to provide multi-resolution

sensing features the wavelet transform has been adopted [3]. Transformation is ap-

plied to the received signal and resulting coefficients are used for representing power
 6

spectral density in a given detection resolution. The power spectral density estimate

can be used to test the non-whiteness of the spectrum.

Three signal specific sensing techniques have been used in signal detection.

These are the signature sequence correlation sensing technique [3], FFT based pilot

sensing [3] and phase-lock loop pilot detection [3].

The ATSC signature sequence correlation sensing technique [7] uses the corre-

lation of the received baseband signal with a signature sequence based on the Data

Field Sync PN sequences. The data field contains a 511-bit PN sequence(PN511) and

three 63-bit PN sequences(PN63). The duration of a single ATSC data frame is 24.2

ms. Multiple ATSC data fields can be used for more sensitive sensing. Test statis-

tics are obtained by processing the correlated output. Detection is accomplished by

comparing test statistics to decision criterion.

The FFT based ATSC pilot sensing is explained in [8]. Non-blind (ATSC-

specific) sensing techniques that meet the sensing sensitivity requirement of IEEE

802.22 uses a 2048-point FFT [3]. This sensing technique involves estimating power

spectral density around a pilot frequency. Several test statistics can be used and one

is using the power spectral density. The power spectral density can be estimated

using the periodogram which is based on the discrete Fourier transform. The ATSC

VSB signal has a pilot at the lower band-edge in a known location relative to the

signal. The pilot frequency is measured from the lower edge of the TV channel. The

unoccupied channel is determined by comparing power estimated around pilot signals

to a predefined threshold amplitude. First the signal is demodulated to baseband.

 7

Then the demodulated signal is filtered with a low-pass filter of adequate bandwidth.

The filter bandwidth should be large enough to accommodate any unknown frequency

offsets. The filtered signal is down-sampled. Finally the FFT of the down-sampled

signal is obtained. The size of the FFT will depend on the dwell time. The maximum

of the squared FFT output value is compared with the threshold value for pilot tone

detection [9].

The PLL based ATSC pilot sensing technique is explained in [3]. Two frequency

tracking blocks are used to track the ATSC pilot frequency. Each of the frequency

trackers are initialized with a tracking frequency close to the expected pilot frequency.

One frequency tracker is initialized with 30 KHz above the pilot frequency and other

is initialized 30 KHz below the pilot frequency. The test statistic is the absolute

value of the difference between the final frequency estimates of each of the frequency

trackers. If this difference in the two final frequency estimates is above a threshold

then the sensing technique decides that an ATSC pilot signal is present [2]. This

sensing technique requires approximately 50 to 75ms. The results depends on the

fading characteristics near the pilot frequency. SNR sensitivity of the technique is

between −12 to −14 dB.

In the wireless microphone method the covariance sensing technique calculates

the covariance matrix of the received signal. If no signal is present that means the

noise only case and this covariance matrix approaches a diagonal matrix since noise

T1
is white. The first test statistic is T = T2
where T1 is the sum of all the magnitude

of the all the elements in the covariance matrix and T2 is the sum the magnitudes
 8

of diagonal elements of the covariance matrix. When the signal is white noise T1

and T2 should be equal since the off diagonal elements should be approximately zero.

When the signal is present the signal and the test statistics for a non-white signal is

no longer close to one [3][10].

In spectral correlation sensing the power spectral density of the received signal

is estimated. This PSD estimate is correlated with a stored PSD of the expected

signal. The test statistic is the correlation of the estimated PSD and prestored PSD

of the signal of interest. If the test statistic exceeds the preset threshold, then the

signal is present.

Cyclostationary feature detection can be applied when cyclostationary signals

are generated by primary users. A cyclostationary signal is a signal that is not

stationary and whose autocorrelation function is periodic. This sensing technique is

explained in [3]. Cyclostationary signal analysis focuses on the ATSC pilot signal.

This signal is filtered with a narrow-band filter. The spectral correlation function is

estimated from the sampled signal. The spectral correlation function is also known

as the cyclic spectrum. The spectral correlation is a function of two parameters:

the frequency index k and cyclic frequency index α. For additive white Gaussian

noise (AWGN), the spectral correlation function is zero for α 6= 0. If the ATSC pilot

signal is present, those terms in the spectral correlation function can become quite

large. The advantages of the cyclostationary detection are robustness to noise and

better detector performance even in very low SNR regions. The disadvantage is a

more complex processing task than energy detection and therefore high speed sensing
 9

cannot be achieved. The method cannot be applied for unknown signals because an

apriori knowledge of the target signal characteristics is needed. At a given time

only one signal can be detected [11][12]. This technique is based on noise having

Gaussian statistics. Higher order statistics can be used for predicting how well the

distribution of the test statistic meets that of the Gaussian hypothesis. Higher order

cumulants are compared with the power of the second order moment to determine if

the received waveform belongs to a DTV signal or noise. This is efficient for detecting

non-Gaussian signals at low SNRs, with noise assumed to be a Gaussian random

process [7][2]. First the received signal is converted to the baseband frequency. The

ATSC pilot frequency is shifted back to DC value and the pilot tone is filtered by a

narrow-band filter. If the signal fails the Gaussian test then the sensing techniques

decides the ATSC pilot signal is present. For Gaussian noise, higher order cumulants

are equal to zero. If only noise is present then the real and imaginary part of the

frequency domain signal are both Gaussian.

1.3 Problem Statement

The transmission and the reception of a narrowband signal using a soundcard

is described for pilot tone detection in spectrum sensing. Pilot tone and energy

detection can be made to provide robust sensing at low SNR and with short sensing

times. Robust and reliable spectrum sensing has a critical role in the dynamical

access of underutilized spectrum. As well as regulatory authority of spectrum (FCC)

changed the rules to allow secondary users for accessing the TV white space spectrum

between 50 − 700M Hz. In order to access TV white space, the secondary system
 10

needs to be able to detect the presence of primary signals at received power levels of

-114 DBm with SNR of -15 dB [13]. Pilot tone and energy detection which tries to

detect the signal presence through comparing the received energy around the pilot

tone against a certain threshold energy. Energy of the pilot tone is 11.3 dB above

pilot tone in the ATSC television transmission. The received intermediate frequency

is converted to a baseband signal and passed through a lowpass filter to retrieve the

low-frequency components. A phase locked loop is used for phase synchronization.

Phase synchronization is challenging in pilot tone detection because the difference

in the phase of the signal affects the magnitude of the signal. In ATSC digital

TV transmission in the United States, the modulated signal occupies the entire 6

MHz TV channel uniformly, with pilot tones between frequencies, 309440.6 Hz and

328843.6 Hz. The frequency is measured from the lower edge of the TV channel.

The difference between these two frequencies is approximately 19.4 KHz. Sound

card is used to input and output the signal to computer for processing. Sound card

does not pass through DC signal. so that pilot tone is filtered out. Transmission

and reception of a narrowband signal using a sound card is analyzed for pilot tone

detection and pdf of the received base band signal is compared with two pdf. One

is pdf of energy of the noise and other is pdf of the energy of signal plus noise. If

the energy of the received signal is greater than a threshold value, one can conclude

that signal is present. If the signal energy is less than the threshold value one can

conclude that signal is absent. Probability of error detection and probability of false

alarm need to be calculated to get the optimum threshold value.

 CHAPTER 2

BASEBAND SIGNAL PROCESSING AND ENERGY DETECTION

This chapter presents an analysis of the statistics of the baseband signal at

the receiver under the influence of low pass filtering. In particular, the statistical

characteristics of time-sampled signals will be examined. The statistics of the noise

only and the signal plus noise case are separately presented in Sections 2.1 and 2.2

respectively. Section 2.3 presents results from simulation that validate the statistical

analysis.

2.1 Statistics of Low-pass Filtered White Noise

Consider a discrete-time noise signal B(i Ts ) sampled at time intervals equal

to Ts where the variable i is a integral valued index. The signal B is passed through

an ideal low-pass filter H(ω) as shown in Fig. 2.1. The filtered output is represented

as n(t) and this signal has a bandwidth equal to 2a radians per second. The signal

n(t) is a continuous time noise waveform.

The output of the filter, n(t) is the result of the convolution of the impulse

response of the low-pass filter h(t) with the time sampled signal B(i Ts ). For the

ideal low pass filter, with bandwidth 2a radians per second, the impulse response of
 12
H(ω)
1

n(t)
B(iTS )
−a a

Figure 2.1: Interpolation of discrete white noise using low-pass filter

the filter can be calculated by taking the inverse Fourier transform of the transfer

function H( ω). The impulse response can be obtained as follows.

a
1 sin(a t)
Z
jωt
h(t) = e dω = (2.1)
2π −a πt

sin(x)
Representing h(t) in the form of a sinc function that is of the form sinc(x) = x
,

the impulse response h(t) can be represented as,

a
h(t) = sinc (a t). (2.2)
π

Therefore the output of the filter is

a X

n(t) = sinc a(t − i Ts ) B(i Ts ) (2.3)
π i

If the sampling frequency of the discrete signal is chosen such that a Ts = π Eqn 2.3

becomes
a X

n(t) = sinc π(t/Ts − i) B(i Ts ) (2.4)
π i

This above summation can be evaluated using the property of the sinc function.

The value of sinc(t) function is equal to one at t = 0 and is equal to zero when t

is an integral multiple of π. The function n(t) represents interpolation of the noise

 13

samples B(i Ts ). The noise samples can be recovered by evaluating n(t) at t = iTs ,

that is,

n(iTs ) = B(i Ts ) (2.5)

a
In Eqn. 2.3, let π
B(i Ts ) = γi . The square of the noise signal,

X X
n2 (t) = sinc (a(t − iTs )) γi sinc (a(t − k Ts )) γk (2.6)
i k

The energy of the noise signal with aTs = π is given by

t t
Z ∞ Z ∞ X X
2


n (t) dt = sinc π( − i) γi sinc π( − k)) γk dt (2.7)
−∞ −∞ i
Ts k
Ts

Interchanging the order of integration and summation in Eqn. 2.7, the expression for

the noise energy can be rewritten as

t t
Z ∞ X X Z ∞
2



n (t)dt = γi γk sinc π( − k) sinc π( − i) dt (2.8)
−∞ i k −∞ Ts Ts

Employing the orthogonal property of the sinc product, the expression for the noise

energy is reduced to,

π
Z ∞ X
n2 (t)dt = γi2 (2.9)
−∞ i
a

Therefore the noise energy per unit time

(N −1)Ts N −1
1 π X 2
Z
2
n (t)dt = γ (2.10)
N Ts 0 N a i=0 i

If each sample γi is an independent normal random variable with zero mean, the sum

of the squares follows the chi-squared distribution with N degrees of freedom [14].

Consider a random variable γ which represents a Gaussian white noise with

mean zero and standard deviation one. The probability density function (pdf) of the
 14

random variable γ is given by

1 −γ 2
fγ (γ) = √ e 2 (2.11)
2π

The square of this noise variable γ will be equated to the random variable z. Hence

z = γ 2 . The probability density function of z can be shown to be [15],

1 √ √

fz (z) = √ fγ ( z) + fγ (− z) (2.12)
2 z

Substituting the pdf of fγ (γ) in Eqn. 2.12 yields the chi-square distribution.

1 −z
fz (z) = √ e2 (2.13)
2zπ

Now, consider the sum of the noise energy after 2M trials. The outcome of the

k th trial is denoted by the random variable zk . The trials are statistically independent

and identically distributed and the random variable zk is taken to be drawn from
P2M
the chi-square distribution. The sum is denoted by Q = k=1 zk . The pdf of Q

denoted fQ (q) can be calculated as the 2M fold convolution of the pdf fz (z). Using

the Laplace transform of fz the Laplace transform of fQ (q) is

2M 1
fQ (s) = fz (s) = (2.14)
2M (s + 12 )M

The inverse Laplace’s transform of Eqn. 2.14 [16] yields the pdf of Q which is the

chi-square distribution with 2M degrees of freedom.

q M −1 − 2q
fQ (q) = e (2.15)
2M (M − 1)!

The mean and variance of the distribution is 2M and 4M respectively. The noise

energy pdf is shown in Fig. 2.2 for the case 2M = (4, 6, 8, 10). This density function

is also referred as the chi-square distribution with M degrees of freedom.

 15
0.2
A
0.18

0.16

0.14
B
0.12
C
fQ(q)

0.1
D

0.08

0.06

0.04

0.02

0
0 5 10 15 20 25 30
q

Figure 2.2: Probability density function of noise energy: (A) 2M = 4, (B) 2M = 6,

(C) 2M = 8, (D) 2M = 10

2.2 Statistics of Low-pass Filtered Signal + Noise

Consider the sampled signal x(i Ts ) = s(i Ts ) + n(i Ts ) where s(i Ts ) is a

random signal and n(i Ts ) represents samples of noise. The received signal x(i Ts )

is passed through a low-pass filter with transfer function H(ω) having the impulse

response


h(t) = sinc a(t − i Ts ) (2.16)

The filtered output is

X


y(t) = sinc a(t − i Ts ) s(i Ts ) + n(i Ts ) (2.17)
i
 16

The instantaneous energy of the signal is obtained by squaring y(t).

XX
y 2 (t) =



sinc(a(t − i Ts ) sinc a(t − k Ts ) s(i Ts )s(k Ts )
i k
XX


+ sinc(a(t − i Ts ) sinc a(t − k Ts ) s(i Ts )n(k Ts )
i k
XX


+ sinc(a(t − i Ts ) sinc a(t − k Ts ) n(i Ts )s(k Ts )
i k
XX


+ sinc(a(t − i Ts ) sinc a(t − k Ts ) n(i Ts )n(k Ts )
i k

(2.18)

The total energy in the signal is given by,

Z ∞ Z ∞ XX
2



y (t)dt = sinc a(t − i Ts ) sinc a(t − k Ts ) s(i Ts )s(k Ts )dt
−∞ −∞ i k
XX


+ sinc a(t − i Ts ) sinc a(t − k Ts ) s(i Ts )n(k Ts )dt
i k
XX


+ sinc a(t − i Ts ) sinc a(t − k Ts ) n(i Ts )s(k Ts )dt
i k
XX


+ sinc a(t − i Ts ) sinc a(t − k Ts ) n(i Ts )n(k Ts )dt
i k

(2.19)

Using the orthogonal property of the sinc function and assuming s and n are uncor-

related, it can be shown that,

πX
Z ∞
2
y 2 (t)dt = s(i Ts ) + n(i Ts ) (2.20)
−∞ a i

In the analysis to follow the signal s(t) is taken to be a binary phase shift key

(BPSK) signal. The binary input (0, 1) is transmitted as a pulse having the ampli-

tude (−A, A) respectively. Each outcome is taken to occur with equal probability.

Hence the probability of transmitting the symbol A or −A is equal to 1/2. The

 17

corresponding pdf of the signal is

1 1
fS (s) = δ(s − A) + δ(s + A) (2.21)
2 2

The pdf of the signal plus noise can be obtained by convolving the pdf of the signal

with pdf of the noise. This function is a mixture of two Gaussian distributions with

mean values of +A and −A respectively. It can be represented as,

1 (y−A)2 (y+A)2

fy (y) = √ e− 2 + e − 2 (2.22)
2 2π

Fig. 2.3 shows an example of the pdf of a BPSK modulated signal with additive

Gaussian white noise. For this case, A = 75 and the noise variance is σ 2 = 1. The

signal to noise ratio SN R = 10 log10 (A2 /2σ 2 ) = 35dB.

0.1

0.09

0.08

0.07

0.06
fy(y)

0.05

0.04

0.03

0.02

0.01

0
-100 -50 0 50 100
y

Figure 2.3: Probability density function of signal + noise

The pdf of the instantaneous energy of the signal, that is z = y 2 can be

calculated by substituting fy (y) in Eqn. 2.12. The result is ,

1 ( z−A)2
√
( z+A)2
√
(− z+A)2
√
(− z+A)2

√

fZ (z) = √ e− 2 + e− 2 + e− 2 + e− 2 (2.23)
4 2πz
 18

z+A 2
Factoring √1 e− 2 from Eqn. 2.23,
4 2πz

1 z+A2 √ √ √ √

fz (z) = √ e− 2 e zA + e− zA + e− zA + e zA (2.24)
4 2πz
Recognizing that,
√ √ √
2 cosh( zA) = e zA + e− zA

The pdf of z is the non-central chi square distribution

1 z+λ̂
p
fz (z) = √ e− 2 cosh( z λ̂) (2.25)
2πz

where the non-centrality parameter λ̂ is equal to A2 . The Laplace transform of fz is

p
e−λ̂/2 ∞ cosh( z λ̂) (s+1/2)z e−λ̂/2
Z
F (s) = √ √ e dz = √ eλ̂/(4ŝ)
2π 0 z 2ŝ

where ŝ = s + 1/2.
P2M
The energy in 2M trials is equal to Q = k=1 zk . The random variables zk are

statistically independent and identically distributed, where zk is drawn from the pdf

given in Eqn. 2.25, the non-central chi square distribution. The Laplace transform

of fQ is equal to the 2M fold product of F (s).

e−λ/2 λ/(4ŝ)
FQ (ŝ) = e
2M ŝM

where ŝ = s + 1/2 and λ = 2M λ̂. Taking inverse Laplace transform and applying

the change of variable s = ŝ − 1/2 yields

e−q/2 i∞+c
Z
fQ (q) = FQ (ŝ)eŝq dŝ
2πi −i∞+c

The aforementioned integral can be evaluated using Eqn (29.3.81) on page 1026 in

Abramowitz and Stegun[17].

ek/ŝ (M −1)/2
p
→ (q/k) I M −1 (2 kq)
ŝM
 19

In our case k = λ/4. The pdf of Q is described by the non-central chi square

distribution with 2M degrees of freedom and its non-centrality parameter is equal

to λ.
  M2−1
1 − q+λ q p
fQ (q) = e 2 IM −1 ( qλ) (2.26)
2 λ

where function IM −1 is the (M − 1)th order modified Bessel function of the first kind

and λ = 2M A2 . The probability density function of Q is shown in the Fig. 2.4.

0.08
A
0.07

0.06

0.05 B
fQ(q)

0.04
C
0.03

0.02

0.01

0
0 20 40 60 80 100 120
Signal + Noise Energy, q

Figure 2.4: Pdf of signal plus noise energy: (A) 2M = 2, (B) 2M = 4, (C) 2M = 8.
Continuous line is FFT based estimation of fQ (q). Symbols are simulated results.

The figure. 2.4 shows the pdf of the simulated result and numerical result of sig-

nal plus noise energy for different window sizes. The numerical result is represented

by the M fold convolution of Eqn. 2.25. Since convolution operation is equivalent to

multiplication operation in Fourier domain, the Fourier transform is first numerically

calculated and the result multiplied M times. Taking the Fourier inverse of the M
 20

fold multiplication of fZ (z) the numerical estimate of fQ (q) is obtained. As the win-

dow size increases the mean and variance of the non-central chi-square distribution

increases.

2.3 Energy Based Signal Detection

Signal detection is based on testing the statistics of the received signal. Two

outcomes are possible. The first outcome is that no signal is received and the second

is that a signal is present. In either case noise will be present. The decision follows a

testing of two hypotheses [14]. The first hypotheses is H0 where the received signal

energy Yn is assumed to be energy of noise having mean zero and variance σ 2

H0 : Y n = W n (2.27)

The second hypothesis H1 is that the received signal energy Yn is a combination of

a signal plus noise.

H1 : Y n = S n + W n (2.28)

The energy of the signal Yn is evaluated as the 2M term sum of the squares of

its samples. Under hypotheses H0 , the pdf is given by the non-central chi-square

distribution with 2M degrees of freedom. The probability density function of the

test statistic under hypotheses H0 is shown in the Fig. 2.2. The objective is to

determine the probability of error and probability of false alarm.

Under the hypothesis H1 , the sum of the energy has been shown to have a non-

central chi-square distribution. The ratio of the signal energy to noise spectral density

provides the signal to noise ratio(SNR). As the amplitude of the signal increases
 21

the signal to noise ratio increases and the pdfs separate, minimizing the region of

overlap. This results in a lowering the probability of detection energy. As the signal

to noise ratio decreases the two pdfs move closer and probability region of overlap

will increase. A threshold value Λ can be calculated that minimizes the probability

of detection error.

The probability of misclassification Pe given the decision threshold value Λ is

Pe = P rob(q ≥ Λ|H0 )P rob(H0 ) + P rob(q ≤ Λ|H1 )P rob(H1 ) (2.29)

where
q M −1 γ(M, Λ/2)
Z ∞
− 2q
P rob(q ≥ Λ|H0 ) = M
e dq = 1 − (2.30)
Λ 2 (M − 1)! (M − 1)!
  (M2−1)
Z Λ
1 − q+λ q p √ √
P rob(q ≤ Λ|H1 ) = e 2 IM −1 ( qλ)dq = 1 − QM ( λ, Λ) (2.31)
0 2 λ

The function γ(M, u) represents the lower incomplete gamma function and QM (a, b)

is the Marcum Q-function. The P rob(H0 ) and P rob(H1 ) represent the probability

of noise and noise plus signal being transmitted respectively. In the results shows

1
that the duty cycle of the signal to noise is 50%, and therefore P rob(H0 ) = 2
and

P rob(H1 ) = 21 . This means that during a time period T , half of the time signal is

transmitted and other half signal is not transmitted. The probability p e versus Λ

is shown in fig. 2.5. The optimum threshold is calculated for a fixed window size

and amplitude of the signal A2 = 3 and window size 2M = 4 as shown in Fig. 2.5.

The optimum threshold value is equal to the value A corresponding to the minimum

value of Pe . The probability of error decreases as the threshold increases. It will

reach an optimum value, after that value error increases as threshold increases.
 22
0.5

0.45

0.4
Probability

0.35

0.3

0.25

0.2
0 5 10 15 20 25 30
Threshold

Figure 2.5: Optimum threshold for 50% duty cycle for M = 4 , A2 = 3 and σ 2 = 1

Next the probabilities of detection error and false alarm are estimated using

computed pdfs. Let Pe represent the probability of detection error and Pf a represent

probability of false alarm. It can be determined as

q M −1
Z ∞ Z ∞
q
Perr = P rob(q ≥ Λ|H1 ) = M
e− 2 dq (2.32)
Λ Λ 2 (M − 1)!

  M2−1
1 − q+λ q
Z ∞ Z ∞ p
Pf a = P rob(q ≥ Λ|H0 ) = e 2 IM −1 ( qλ) dq (2.33)
Λ Λ 2 λ

The error probability and false alarm probability for the signal plus noise case

considering A2 = 4 is given in Fig.2.6. A2 = 1 and σ 2 = 1 is given in Fig. 2.7.

In Fig.2.6 represents the probabilities of false alarm pf a and probabilities of de-

 23
70

60

50

40
Threshold

30 2M=8

20

2M=4

10
2M=2

0
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Probability

Figure 2.6: The error probability and the probability of false alarm for SN R = 4

tection error perr as a function of threshold parameter on the vertical axis. The differ-

ent window sizes are considered. The false alarm probability increases as the thresh-

old decreases, where as the probability of detection error increases as the threshold

increases. Based on performance requirements, a threshold can be selected from

these results. For example, the probability at which the two curves intersect, yields

an optimal threshold value that provides the smallest false alarm and detection error

probabilities. If one requires a smaller detection error probability, a smaller threshold

will be required at the cost of higher false alarm probability. In Fig.2.6, it is seen

that for SNR 1 selecting such a threshold results in a increase in a probability of

detection from 76% to 93% as the window size increase from 2M = 2 to 2M = 8 .

The obtained probability of detection ( Pdet ) and probability of false alarm ( Pf a )

 24
Position 2M Threshold Pdet Pfa
1 2 8 0.76 0.24
2 4 12 0.84 0.16
3 8 25 0.93 0.07

Table 2.1: The probability of detection ( Pdet ) and the probability of false alarm
( Pf a ) for SN R = 4

for SN R = 4 is shown in Table. 2.3

35

30

25

20 2M=8
Threshold

15

10
2M=4

2M=2
0
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Probability

Figure 2.7: The error probability and the false alarm probability of signal plus noise
for SN R = 1

In Fig.2.7, results are shown in for case when SNR is decreased to 1 . There

is no significant increase in probability of detection for the selected window sizes.

As SNR decreases, much larger windows are required for probability of detection in

the 90% range. The energy detection method is not suitable for low SNR regimes.
 25
Position 2M Threshold Pdet Pfa
1 2 8 0.52 0.48
2 4 12 0.55 0.45
3 8 25 0.57 0.43

Table 2.2: The probability of detection ( Pdet ) and the probability of false alarm
( Pf a ) for SN R = 1

Obtained probability of detection (Pdet) and probability of false alarm (pfa) for SN R =

1 is shown in Table. 2.3

 CHAPTER 3

DETECTION OF BASEBAND SIGNAL

3.1 Introduction

Digital TV in the United States operates according to the Advanced Televi-

sion Systems Committee (ATSC) standards. TV signal broadcasting takes place

on a channel that is 6 MHz wide. The video signals are modulated onto the radio

frequency (RF) carrier using the 8 Vestigial Side Band (VSB) technique. The RF

carrier can range from 30 KHz to 300 GHz. The video transmission involves a

sequence of stages of processing the MPEG-2 input video stream that includes ran-

domizing the data, Reed-Solomon encoding, data interleaving and trellis encoding.

The resulting signal uniformly occupies most the entire 6 MHz channel. This sig-

nal is then multiplexed with segments for synchronization and recovery of the data.

Prior to VSB modulation, pilot carrier signals are added to the data signal. This

is done by applying a DC offset of 1.25 V to the zero mean baseband data signal.

This results in a carrier signal appearing at the lower edge of the transmission band.

In particular, the ATSC pilot appears at the frequency of 310 KHz from the lower

band edge and contains 7% of the total signal power, contributing 0.3 dB to the total

signal power. The resulting data plus pilot signal undergoes VSB modulation, using
 27

a single sideband suppressed carrier scheme [18].

The digital signal is split in two and passed through a raised cosine filter and

processed by a digital to analog converter. The process of generating the VSB signal

involves amplitude modulating the input signal and passing the result through a filter.

The filter is used to partially remove the lower sideband of the modulated signal. The

process yields a signal modulated at an intermediate frequency 44 M Hz. Following

this stage, the signals are up-converted to the required RF carrier frequency and

then passed through a power amplifier and the transmit antenna. A schematic of

the spectrum of the 8VSB signal is shown in Fig. 3.1.

Figure 3.1: Spectrum of VSB (http://www.8vsb.com)

[REF: http://www.8vsb.com].

3.2 Problem Statement

A band-limited narrow-band signal is generated using a signal generator and

the ATSC transmission and reception of signal for TV video is analyzed for two
 28

channels using a sound-card. The detection of pilot tone for spectrum sensing is

investigated. The band limited information signal and modulated version of the

narrow-band information signal are added together to yield a single signal for trans-

mission. The transmitting and receiving signal is processed through a sound card.

The sound card facilitates input and output of the signal to and from a computer

under control of portaudio (http://portaudio.com) software. Portaudio is a free,

cross platform, open source audio I/O library. It allows one to write simple audio

programs in C/C++. The received signal is passed through a lowpass filter to yield

the narrow-band signal back. The received signal is passed through highpass filter

and demodulated using a phase locked loop (PLL) to yield the second channel. AM

modulation is considered for simplicity, since the main objective is in the application

of the pilot tone for sensing the signal.

3.3 Signal Processing of Transmitted Signal

A baseband signal is modulated to intermediate frequency to allow one to

accommodate more channels. There are different classifications of modulation. First

classification is AM (amplitude modulation) and FM (frequency modulation). The

AM modulation is further divided into double sideband amplitude modulation and

single sideband amplitude modulation. In double sideband AM modulation frequency

span of the signal is doubled in frequency after modulation. The sidebands containing

frequency higher than carrier frequency are called upper sidebands, and those lower

are called lower sidebands. Both sidebands contain equivalent information. Hence

only one sideband is required to recover the input signal without error. This single
 29

sideband modulation efficiently uses the electrical power and bandwidth. Single

sideband modulation avoids bandwidth doubling and power wasting on a carrier at

the cost of device complexity.

In this section, a band-limited signal and the pilot tone are simulated so as to

represent the general features of the 8VSB video signal. Amplitude modulation is

considered for simplicity, since the main objective is in the application of the pilot

tone for sensing the signal.

AM Modulation In AM modulation the amplitude of the carrier signal is

varied in accordance to the amplitude of the information signal. Let xs (t) represent

the transmitted baseband signal and xc (t) = cos(ωc t) represent the sinusoidal carrier,

where ωc /(2π) = fc is the carrier frequency. The suppressed carrier modulated signal

xm (t) = xs (t)xc (t). The Fourier transform of xm (t) is

1
Xm (ω) = (Xs (ω − ωc )) + Xs (ω + ωc ))) . (3.1)
2

where Xs represents the Fourier transform of xs (t). The passband of the information

signal xs (t) has a spectrum Xs (ω) with a frequency response as shown in Figure 3.2.

It has a maximum angular frequency of 104 π rad/s and a bandwidth of 2×104 π rad/s.

The carrier ωc is selected as 2 × 104 π rad/s. The frequency response Xm (ω) of the

amplitude modulated signal is shown in Figure 3.3.

Amplitude modulation is governed by the modulation index m, which can be

defined as the ratio of the peak value of modulating signal xm (t) to the peak value

of the carrier signal, m. It is a measure of the extent of amplitude variation of the

 30

Xs (ω)

−10π 0 10π
ω
1000
rad/s

Figure 3.2: Passband spectrum of the information signal

Xm (ω)

1
2

omega
1000
−30π − 20π − 10π 0 10π 20π 30π

Figure 3.3: Modulated suppressed carrier passband signal

modulated signal about the modulated carrier signal and affects the ability of the

receiver to accurately demodulate the signal. Consider the transmitted AM signal

xtrans

xtrans (t) = xc (t) + mxs (t)xc (t) (3.2)

where |xs (t)| < 1. If m < 1 signal can be demodulated without distortion.The

temporal variation of the modulated signal for this case is shown in Fig. 3.4 where

xs is a single sinusoid having a frequency less than ωc . In this case the envelope of

the modulated signal is equivalent to that of the baseband signal. For values of the

modulation index greater than one, the envelope becomes distorted and may cause
 31

interference to the other users. The effect is shown in Fig. 3.5. In such a case the

modulated carrier signal periodically reduces to zero resulting in drop outs of the

information signal at the receiving end. If m ≥ 1 the signal cannot be recovered

accurately.

1.5

0.5
Amplitude

-0.5

-1

-1.5
0 0.0005 0.001 0.0015 0.002 0.0025 0.003
Time(sec)

Figure 3.4: Modulated signal when modulation index, m ≤ 1

1.5

0.5
Amplitude

-0.5

-1

-1.5

-2
0 0.0005 0.001 0.0015 0.002 0.0025 0.003
Time(sec)

Figure 3.5: Modulated signal when modulation index, m ≥ 1

 32

At the receiver, the signal recorded by the antenna is passed through a low noise

amplifier and then a mixer is used to down convert the RF signal to an IF signal.

The angular frequency of the receiver oscillator must be exactly the same as that of

the transmitted carrier ωc . If this is not the case, it results in a angular frequency

shift ωc + δω, causes distortion in the receiver signal. Before demodulation, it is very

important to check whether the phase of the signal has been altered. If the receiver

and transmitter oscillators are not in phase the amplitude of the demodulated signal

amplitude will be reduced by a factor of the sine of the phase angle difference. A

phase locked loop is required to synchronize the phase between transmitter and the

receiver.

After modulation, the amplitude of the modulated signal spectrum is half of

the magnitude of the original signal. The output signal is the shifted version of

the input with two parts, one is centered at 2 × 104 π rad/s and other is centered at

−2×104 π rad/s. The signal used for transmission is the sum of the input signal xs (t)

and the frequency shifted version of input signal resulting from modulation which is

xm (t). These two signals are added to obtain a signal x(t) = xs (t) + 2 xm (t) with

a frequency bandwidth approximately 6 × 104 πrad/s. A pilot tone xp (t) = sin(ωp t)

is then also added to the transmitting signal x(t). Power level of the pilot tone is 11

dB above the power level of the signal. The resulting signal in the frequency domain

is shown in Figure 3.6.

Finally, the signal needs to be up-converted to a radio frequency to be suitable

for transmission. A radio frequency of 2.4 GHz is used for transmission. Pilot signals
 33

X(ω)

ω
−ω 1000
−30π − 20π − 10π 0 10π 20π 30π

Figure 3.6: Baseband signal

are added to the RF modulated signal. The spectrum of the RF modulated signal

with pilot tones incorporated is shown in Figure 3.7.

Xt (ω)
A − − − − −|∆ω|
−−−

B−−−−

ωol ωh ω

Figure 3.7: Transmitted signal

The next section discusses the important elements in the processing of the

received signal.
 34

3.4 Signal Processing of Received Signal

The transmitted RF signal is captured using a 2.4GHz RF receiver and trans-

mitted signal recovered. The received signal is processed through an external sound

card connected to USB port of a PC. The sound card also known as the audio card

facilitates input and output of the signal to and from a computer under the control

of a computer program. Since the sound card does not pass DC components, the

pilot tones are filtered out.

The received signal is first passed through a phase locked loop (PLL) and then

demodulate to recover baseband frequency signal. A schematic of the recovered

signal after demodulation is shown in Figure 3.8.

0 ωh − ω o l ω

Figure 3.8: Received signal after demodulation

The angular frequency of the receiver oscillator must be exactly the same as

that of the transmitted carrier which is ωc = 2 × 104 π rad/sec. If not, this results

in a frequency shift ωc + δω), that causes distortion. A phase locked loop is used
 35

synchronize the transmitted and received signals. The next section discusses in detail

the operation of the phase locked loop (PLL).

Phase Synchronization Using a Phase Locked Loop

Television communication systems have to recover the modulated radio fre-

quency signal. A phase locked loop (PLL) is a closed loop control system with an

oscillator to provide an output signal that maintains a constant phase angle with

reference to a reference signal. During the transmission of an amplitude modulated

signal, the frequency and phase of the transmitted signal is distorted due to noise

and interference. It is required to ensure that the receiver’s carrier frequency and

phase are same as that of the transmitted signal. The PLL ensures both transmitted

and received carrier signals are in phase.

An amplitude modulated signal defined by the equation

x(t) = sin(ωc t + φ) (3.3)

where ωc is the carrier frequency and φ is the phase angle.

A block diagram of the PLL is shown in Figure 3.10. The PLL consists of two

signal multipliers, two lowpass filters, a −900 phase shifter, a voltage controlled oscil-

lator (VCO) and an integrator. The received signal x(t) = sin(ωt + φ) is multiplied

with two separate signals e1 (t) = cos(ωt + θ), e2 (t) = sin(ωt + θ) using the two

mixers. One is the direct signal output of the VCO. The other is the 900 shifted sig-

1
nal of the VCO output. The outputs of the mixers are v1 (t) = 2
sin( 2ω t +

1
φ + θ) + sin(φ − θ) and v2 (t) = 2
cos(2ωt + φ + θ) + cos(φ − θ). The low
 36

Figure 3.9: Block diagram of phase locked loop

frequency component of the output of the multipliers is a signal whose magni-

tude and sign are proportional to the phase difference for small differences be-

tween input signal and output of the VCO. These signals are passed through low-

pass filters to remove the frequency component of 2 ω that is generated by signal

multiplication. The output of the two filters are represented as f1 (t) and f2 (t) ,

1 1
where f1 (t) = 2
sin(φ − θ) and f2 (t) = 2
cos(φ − θ). The output of the filters are

1 1
f1 and f2 , where f1 = 2
sin(φ − θ) and f2 = 2
cos(φ − θ). Taking arctangent of

f1 /f2 yields φ − θ. The control loop attempts to keep the phase difference between

VCO output and input signal small. The error φ − θ is used to correct the phase
 37

difference of the input and the feedback components. The use of the arctangent

eliminates the need for amplitude scaling of the signals. This process allows one to

synchronize the oscillator with the input signal.

0.45

0.4

0.35
theta

0.3

0.25

0.2

0.15
0 50 100 150 200
Iteration count

Figure 3.10: Theta vs N

Implementation of the Phase Locked Loop

The PLL was implemented as a software module. The received baseband signal

is used for processing. The received signal is processed through sound card of the PC.

The sound card facilitates input and output of the signal to and from the PC under

control of computer program. The sound card does not pass through DC. So the

pilot tone is filtered out. The input signals are generated using a signal generator,

Bruel and Kjaer’s dual channel signal analyzer 2032 series and using a bandpass

filter. The cutoff frequencies are 0 and 5 KHz.

 38

The carrier signal is generated using a sine wave generator with frequency

10 KHz. The signal is multiplied with the carrier signal using a Analog Devices

AD632 signal multiplier. The signal is transmitted and received using a 2.4 GHz

transmitter and receiver. The output of PLL for a input sin(ωt + φ) is shown in

Fig. 3.10, where φ = π/8 and θ is synchronized to φ. The next section describes the

method used for designing the filters in the PLL.

Design of Butterworth Filters Using Bilinear Transformation

In signal processing, the function of a filter is to remove unwanted frequency

components and random noise from a signal or to extract useful parts of the signal

residing in a certain frequency range. The Butterworth lowpass filter is characterized

by a flat frequency response in the passband. The squared magnitude of the transfer

function of an analog Butterworth filter is of the form

1
|H(ω)|2 = (3.4)
1 + ( ωωc )2N

where N is the filter order and ωc is the cutoff frequency. The bilinear transfor-

mation can be used to obtain the infinite impulse response realization of the digital

filter from the analog filter response. The bilinear transformation is defined as
 
1 − z −1
s= C (3.5)
1 + z −1

2
where C = Ts
and Ts is the sampling time.

Design of the Lowpass Filter

A lowpass filter is a linear time invariant system (LTI) that admits the low
 39

frequency components well and attenuates frequencies higher than the filter’s cutoff

frequency. The second order Butterworth lowpass filter has the Laplace transform

1
H(s) = √ (3.6)
s2 + 2s + 1

Substituting Eqn. 3.4 into Eqn 3.5 yields

Y (z) z −2 + 2 z −1 + 1
= √ √ (3.7)
X(z) (C 2 − 2 C + 1)z −2 + (2 − 2 C 2 )z −1 + C 2 + 2 C + 1

Cross multiplying Eqn. 3.7.

√
Y (z)( (C 2 − 2C + 1)z −2 + (2 − 2 C 2 )z −1
√
+ (C 2 + 2 C + 1)) = X(z)(z −2 + 2z −1 + 1) (3.8)

Taking inverse z-transform

√
y(n − 2)(C 2 − 2C + 1) + (2 − 2C 2 )y(n − 1) +
√
(C 2 + 2C + 1)y(n) = x(n − 2) + 2x(n − 1) + x(n) (3.9)

Hence

y(n) = a0 x(n) + a1 x(n − 1) + a2 x(n − 2) − b1 y(n − 1) − b2 y(n − 2) (3.10)

a0 =1 (2−C 2 )
√
C 2 − 2 C+1
√
where Arg
, a1 = 2a0 , a2 = a0 , b1 = Arg
, b2 = Arg
and Arg = C 2 + 2C + 1

Design of the Highpass Filter A highpass filter is a linear time invariant

system that passes high frequency components and attenuates frequencies lower than
 40

the filter’s cutoff frequency. The second order highpass filter can be obtained for given

1
lowpass filter, by substituting s
for s. The second order Butterworth highpass filter

is given by
s2
H(s) = √ (3.11)
s2 + s 2 + 1

Using the bilinear transformation

Y (z) C 2 (1 − z−1)2
= √ √ (3.12)
X(z) (1 + C 2 − 2 C)z −2 + (2 − 2 C 2 )z −1 + (1 + 2 C + C 2 )
√
Dividing numerator and denominator of the equation by arg = 1 + 2 C + C 2 we

obtain the equation in the form

C 2
Y (z) arg
(1 − 2z −1 + z −2 )
= √ (3.13)
X(z) (1 − 2 C + C 2 ) zarg + (2 − 2C 2 ) zarg + 1
−2 −1

Cross multiplying and taking the inverse z-transform, Equation 3.13 can be rewritten

as

y(n) = a0 x(n) + a1 x(n − 1) + a2 x(n − 2)) − b2 y(n − 2) − b1 y(n − 1) (3.14)

√
C2 (2−2C 2 ) 1− 2C+C 2
where a0 = arg
, a1 = −2a0 , a2 = a0 , b1 = arg
, and b2 = arg
.

Implementation of the High and Low-pass Filters The highpass and

lowpass filters were implemented as a software module in Fortran using the coeffi-

cients obtained. The received signal is processed through the sound card. A sound

card facilitates input and output of the signal to and from a computer under con-

trol of portaudio software. The sound card does not pass through DC so the lowest

frequency pilot tone will be suppressed.

 41

19KHz
↔
A

0 25KHz ∗ 2π ω
Figure 3.11: Down-converted signal

The signal shown in Figure 3.11 is used for further processing of energy detec-

tion. It is basically 5 × 104 π rad/s bandwidth around the pilot tone. Two pilot tones

are placed 3.8 × 104 π rad/s apart. The baseband received signal is further processed

as follows. To convert this intermediate frequency signal to a low frequency signal,

mixer is used.

3.5 Energy Statistics of Narrowband Noise

Consider the discrete time sampled bandpass noise signal n(iTs ) where Ts is the

sampling time and i is integer valued index. The continuous time signal can be re-

covered using a ideal lowpass filter shown in Figure 3.12. Define y(t) = Re [n(t)e jωc t ]

where n(t) is Gaussian noise with zero mean, unit variance and ωc is the center fre-

quency. The complex noise components nr (t) and ni (t) represent the in-phase and

quadrature phase components respectively. Hence

y(t) = nr (t) cos(ωc t) − ni (t) sin(ωc t) (3.15)

 42

H(ω)
1

n(t)
n(iTS )
−a a

Figure 3.12: Transfer function of lowpass filter

P P
where nr (t) = i nr (iTs )sinc(aTs (t/Ts −i)) and ni (t) = i ni (iTs )sinc(aTs (t/Ts −i)).

In order to find the energy of the signal n(t), it is squared and integrated with

respect to time. When this signal passes through the lowpass filter, higher order

harmonics are removed and only the low frequency components in the signals are

retained.
n2r (t) + n2i (t)
Z ∞ Z ∞
2
y (t) = dt (3.16)
−∞ −∞ 2

Interchanging integration and summation and applying the orthogonality property

of the sinc function yields

Z ∞ X n2 (iTs ) + n2 (iTs )
r i
y 2 (t)dt = (3.17)
−∞ i
2a

Therefore the energy of the noise signal is equal to the sum of the squares of the

in-phase and quadrature samples.

P2M
Consider now consider the sum q = i=1 zk where zk is the energy measured in

the k − th trial. In such a case zk = n2r (kTs ) + n2i (kTs ). The noise samples nr and ni
 43

are Gaussian distributed with zero mean and unit variance. Hence zk is Chi-square

distributed with two degrees of freedom. The corresponding pdf is

1
fZk (zk ) = e−zk /2 (3.18)
2

Assuming that the trials are statistically independent and identically distributed the
P2M
sum q = i=1 zk will be Chi-square with 4M degrees of freedom and the pdf,

q 2M −1
fQ (q) = e−q/2 (3.19)
4M (2M − 1)!

Pdf of the noise energy in the intermediate frequency range is shown in Fig. 3.13 for

different window size which varies from 2M = 2, 4, 8, 10.

0.12
A

0.1

0.08
C
D
fQ(q)

0.06

0.04

0.02

0
0 5 10 15 20 25 30
q

Figure 3.13: Pdf of noise energy in the intermediate frequency: (A) 2M = 2,

(B) 2M = 4, (C) 2M = 8, (D) 2M = 10

3.6 Energy Statistics of Narrowband Signal + Noise

Consider the signal y(t) = Re [(n(t) + s(t))ejωc t ] where ωc is the center fre-

quency. The complex amplitudes s(t) = sr (t) + jsi (t) and n(t) = nr (t) + jni (t)
 44

correspond to information and noise signals respectively. Therefore

y(t) = [nr (t) + sr (t)] cos(ωc t) − [ni (t) + si (t)] sin(ωc t) (3.20)

where
X
nr (t) + sr (t) = [nr (iTs ) + sr (iTs )] sinc(aTs (t/Ts − i)) (3.21)
i
X
ni (t) + si (t) = [ni (iTs ) + si (iTs )] sinc(aTs (t/Ts − i)) (3.22)
i

Retaining only the lowest frequency terms of the signal energy

1
Z ∞ Z ∞
2
y (t)dt = [nr (t) + sr (t)]2 + [ni (t) + si (t)]2 dt (3.23)
−∞ 2 −∞

Applying the orthogonality condition of the sinc function one can simplify the afore-

mentioned expression as
Z ∞ X [nr (iTs ) + sr (iTs )]2 + [ni (iTs ) + si (iTs )]2
y 2 (t)dt = (3.24)
−∞ i
2a
√
Consider the statistics of a single sample of the energy zk = (nr (kTs ) + λ)2 +

n2i (kTs ). In such a case si = 0 and s2r = λ. Expanding the expression for zk it can

be shown that
p
n2r + n2i = zk − 2cos(θ) zk λ + λ (3.25)

where θ is phase angle between in-phase and quadrature components of the signal.

The pdf of zk is equal to the non-central Chi-square distribution with two degrees of

freedom.

π √
1 e−(zk +λ)/2 p
Z
zk −2cos(θ) zk λ+λ
fZk (zk ) = e − 2 dθ = I0 ( zk λ) (3.26)
4π −π 2

where I0 is the 0 − th order modified Bessel function of the first kind. If each energy

sample is statistically independent and is identically distributed it follows that the

 45

P2M
sum of 2M trials, q = k=1 zk will be non-central Chi-square distributed with 4M

degrees of freedom.

e−(q+λ)/2  q  M2−1 p
fQ (q) = IM −1 ( λq) (3.27)
2 λ

where Im is the mth order modified Bessel function of the first kind.3.14
0.09

0.08 A

0.07

0.06

0.05 B
Pdf

0.04
C

0.03

D
0.02

0.01

0
0 10 20 30 40 50 60 70 80 90 100
samples

Figure 3.14: Pdf of the signal plus noise(Analytical and simulation

Method): (A) 2M = 2, (B) 2M = 4, (C) 2M = 6, (D) 2M = 8

3.7 Detection of a Narrow-band Signal in Noise

The detection algorithm is based on two test statistics of received signal, One

is H0 and the other is H1 . In H0 , it is assumed that received signal y(t) is only noise

with mean zero and noise variance σ 2 . The energy of the signal is evaluated for 2M

terms. The pdf of the received signal is a central chi-square distribution with 2M

degrees of freedom. The hypotheses H0 is given by

H0 : y(t) = n(t) (3.28)

 46

In the second hypotheses received signal y(t) is a combination of signal plus

noise.

H1 : y(t) = n(t) + s(t) (3.29)

The energy of the signal is calculated for 2M terms and the pdf of the received

signal under hypotheses H1 is a non-central chi-square distribution with 2M degrees of

freedom. Ultimate aim is to find the error probability and the false alarm probability

of the received signal.

A threshold value Λ can be taken to obtain a minimum probability of error

detection. The probability of error detection Pe is given for a decision threshold Λ is

Pe = P rob(q ≥ Λ|H0 )P rob(H0 ) + P rob(q ≤ Λ|H1 )P rob(H1 ) (3.30)

where

q 2M −1 γ(2M, Λ/2)
Z ∞
q
P rob(q ≥ Λ|H0 ) = M
e− 2 dq = 1 − (3.31)
Λ 4 (2M − 1)! (2M − 1)!
  (M2−1)
Z Λ
1 − q+λ q p √ √
P rob(q ≤ Λ|H1 ) = e 2 IM −1 ( qλ)dq = 1 − Q2M ( λ, Λ) (3.32)
0 2 λ

The function γ(M, u) represents the lower incomplete gamma function and QM (a, b)

is the Marcum Q-function. The P rob(H0 ) and P rob(H1 ) represent the probability

of noise and noise plus signal being transmitted respectively. In the results shows

1
that the duty cycle of the signal to noise is 50%, and therefore P rob(H0 ) = 2
and

P rob(H1 ) = 21 . This means that during a time period T , half of the time signal is

transmitted and other half signal is not transmitted.

Fig. 3.15 shows the probability of error detection for different threshold for

fixed window size. The Fig. 3.15 shows that Minimum error is obtained for window
 47
0.5

0.45

0.4

0.35

Pe
0.3

0.25
E
D
0.2 C

B
0.15

A
0.1
0 5 10 15 20 25 30 35 40
Threshold

Figure 3.15: Pe versus Threshold Amplitude for fixed window size: (A) 2M = 2,
(B) 2M = 4 , (C) 2M = 6, (D) 2M = 8, (E) 2M = 10

size 2M = 2. Keeping this window size fixed, changing signal to noise ratio.

0.1

0.01
Pe

0.001

0.0001

1e-05

1e-06
1 10 100
SNR

Figure 3.16: Pe versus SN R for fixed window size 2M = 2

Fig. 3.16 shows the probability of error detection for different signal to noise

ratio (SNR) for fixed window size. Window size is chosen in such a way that it can
 48

yield minimum probability of error. The probability of error is calculated by varying

signal to noise ratio. The minimum error obtained for 2M = 2 . In the modulated

baseband signal detection probability of error is less when window size and threshold

is less.
 CHAPTER 4

CONCLUSION

A band limited narrow-band signal is generated using a signal generator and

the ATSC transmission, reception of signal for TV video is analyzed for two chan-

nels using a soundcard. The transmitting and receiving signal is processed through

a soundcard. The received signal is passed through a lowpass filter to obtain the

narrowband signal back, it is analyzed and the pdf of the signal energy is calcu-

lated based on two hypotheses. The first hypotheses is the received signal energy

is only noise. The second hypotheses is the received signal energy is a combination

of signal plus noise. In both cases pdf of the received signal energy is calculated,

for the first case in which transmitted signal is noise, the pdf of signal energy is a

central Chi-square distribution with 2M degrees of freedom. For the second case in

which transmitted signal is a combination of signal plus noise, the pdf of signal plus

noise energy is a non-central Chi-square distribution with 2M degrees of freedom.

The probability of error detection and probability of false alarm is determined for

a threshold value. In order to phase synchronize the signal at the receiving end, a

phase locked loop (PLL) is designed and implemented.

The probability of error for signal plus noise changes as parameters window
 50

size and the SN R changes. The probability of detection increases as the window

size increases in fixed SN R value. It does not change significantly in the low

SN R value as the window size increases. As the signal to noise ratio decreases,

much larger windows are required for probability of detection in the higher ranges.
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 BIOGRAPHY

Raji Remany was born on May 18, 1980 at Trivandrum Kerala, India. She

earned her Bachelor degree in Electrical Engineering from University of Kerala in

May,2001. In 2008, she enrolled in the Masters program at the University of Mas-

sachusetts Lowell. She currently works as a research assistant in Center for Advanced

Computation and Telecommunication in University of Massachusetts Lowell.